Hi,

In conventional speech processing applications, speech signal is encoded using fixed number of bits over the entire speech signal band. During the process, the band width requirement for speech transmission is relatively high which is of concern.

The QMF means Quadrature Mirror Filter banks are the fundamental building blocks for spectral splitting. The QMF Structure allows spectral decomposition into contiguous overlapping sub bands in such a way that aliasing incurred in the initial “analysis” stage is eliminated during signal reconstruction by the ‘synthesis’ stage The technique is developed to design the perfect reconstruction QMF bank, which allows complete elimination of amplitude n phase distortion of the reconstructed signal . A QMF bank can be formed from FIR or IIR filters.

The aim of the paper is to design a QMF filter and then pass a speech signal through it. The low pass filtered signal is decimated and encoded with more number of bits and high pass filtered signal is also decimated and encoded with less number of bits. These two bit streams are multiplexed and transmitted .In receiver side the received signal is demultiplexed and decoded. The signal is passed through the interpolation and then through the synthesis filters so as to
reconstruct the speech signal. The reconstructed signal is compared with the original speech signal. 

Multi rate signal processing system:

The basic theory of multirate digital signal processing is introduced in this section along with the two Sampling rate alteration device namely Up sampler & Down sampler . In many practical applications where the signal of a given sampling rate needs to be converted into an equivalent signal with a different sampling rate. For example, in digital audio, three different sampling rates are presently employed: 32 kHz in broad casting, 44.1 kHz in digital compact disk and 48 kHz in digital audio tape (DAT) and other applications. Thus conversion of sampling rates of audio signals between these three different rates is often necessary in many situations.

The Discrete–time systems with unequal sampling rates at various parts of the system are called Multirate systems. Unlike in single rate systems the sampling rates at the input and at the output and all the internal nodes are the same. To achieve different sampling rates at different stages, multirate digital signal processing systems employ the downsampler and the up-sampler, the basic sampling rate alteration devices in addition to the conventional elements such as the adder, the multiplier and the delay. Many multirate systems employ a bank of filters with either a common input or a summed output. 

The two basic components in sampling rate alteration are the up-sampler and the down-sampler. For sampling rate alterations, the basic sampling rate alteration devices are invariably employed together with low pass digital filters.

Upsampler:

An up-sampler is a device, which increases the sampling rate by an integer factor. The up-sampler is also called as a sampling rate expander or simply expander.

Downsampler:

A down-sampler is a device which reduces the sampling rate by an integer factor. The down-sampler is also called as sampling rate compressor.

To sum up, The intention of this work is to design and implement a SUBBAND CODING system. We have successfully designed an optimum low pass filter for Four channel QMF Bank to minimize the amplitude distortion. From this Low pass filter we have designed a High pass Filter. Using these filters we have successfully simulated a two channel QMF bank for subband coding of input speech signal. The result shows that the output is a perfect reconstruction of the input speech signal.

Regards,
Karthik.

• Written by Karthik.
• In category Engineering.
• 14 years ago.

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